In Real Time Protocol (RTP) communications, videoconference servers receive audio packets and associated video packets incoming from a client device in order to route the audio and video packets to another client device. A videoconference server processes the received audio packets, for example by compressing a received audio packet and decompressing received compressed packets or by mixing several audio streams received from several client devices. Therefore, a processing delay is added to the transmission delay of the audio packets. In order to synchronize associated video packets to the audio packets, the processing delay is also applied to the video packets by the videoconference server in order to synchronize the video packets to the audio packets prior to sending the audio and video packets to a client device.
In the case where a video packet is lost in the transmission between an emitting client device and a receiving client device, one of the widely used techniques in real time video conferencing is the packet retransmission mechanism. In this mechanism a recovery request is sent from the receiving client device in order that the emitting client device re-sends the lost packet.
Therefore, the transmission delay for the lost video packet can exceed the maximum delay of lip synchronization in the receiving client device, which leads to the inefficiency of the retransmission mechanism.